cisco unity automated attendant not letting me transfer

[anand.narine1]

I wanted to have an automated attendant answer whenever someone dials my phone number so I setincoming calls to be directed to the voice mail extension 2500.

If i call from a cell phone on the pstn, or from an internal phone connected to fxs port to 2500, I do get the attendant "hello cisco unity messaging system.." and hear the options, but as soon I press 4 to get the directory listing or start to enter an extension number, the attendant unsuccessfully tries to connect me to an operator - "sorry operator is not available". However if i call 2500 from an internally ip phone  the attendant works just fine and I can select the option to transfer to an extension or get someone in the directory.

[anand.narine1]

no i need dtmf-relay on the voice router gateway?

[Jaime Valencia]

Yes, that does sound like there's maybe a DTMF problem in your environment.

What protocol is being used?

HTH

java

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[anand.narine1]

i am new to voice.

the gateway router is set up as a h.323 in cucm.

the ip phones are using sccp. not sure if that helps

[Jaime Valencia]

Post your inbound DPs

HTH

java

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[anand.narine1]

voice-port 2/0/0

  connection plar opx 2500     //(fxo port directs incoming calls to the voicemail number 2500 for automated attendant)//

dial-peer voice 1101 pots

  destination-pattern 1101      //fxs port

  port 2/1/0

!

dial-peer voice 90 pots

  description local calls  //calls outgoing to pstn

  destination-pattern .......

  port 2/0/0

!

dial-peer voice 2000 voip

  destination-pattern 5...

  session target ipv4:192.168.1.60   //my cucm server

[Jaime Valencia · Accepted solution]

If that's all you have it's not surprise it's not working (although nothing should work since there's no DP for 2500 there), there's no DTMF config there in the VOIP DP

Add this:

dtmf-relay h245-alphanumeric

HTH

java

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[anand.narine1]

it was supposed to be

dial-peer voice 2000 voip

  destination-pattern 25..

  session target ipv4:192.168.1.60 

i will try ur suggestion and let you know today. btw do i need to create a call handler ? I didn't. I'm just using the voicemail number that everyone dials in to check their voicemail (2500) as the autoattendant.

[Jaime Valencia]

OK, that makes more sense.

Yes, you should use a separate DN, a CTI RP with CFA to VM, how to configure an AA in CUC has been discussed thoroughly before, do a quick search on CSC to get info on that.

HTH

java

if this helps, please rate

www.cisco.com/go/pdihelpdesk